1. Field of the Invention
The invention relates to a sampling frequency converter for converting the sampling frequency of a discrete-time signal from a first frequency to a second frequency, comprising:
a first clock pulse generator for generating first clock pulses which occur with the first frequency,
a second clock pulse generator for generating second clock pulses which occur with the second frequency,
selecting means for supplying selected samples of the discrete-time signal at the rhythm of third clock pulses which occur with a third frequency, and
a selection signal generator for generating the third clock pulses.
2. Description of the Related Art
A converter of this type is known from "A Universal Digital Sampling Frequency Converter For Digital Audio", Proceedings IEEE International Conference on Acoustics, Speech and Signal Processing 1981 (ICASSP 81), Vol. 2, pp. 595-598, FIG. 1B. Sampling frequency converters are used when the sampling frequency of a digitized signal is to be changed. This occurs especially with digital audio signals. In the Compact Disc system, the sampling frequency is 44.1 Khz, in satellite broadcasting, 32 Khz, and in professional recording, 48 Khz. If digital signal sources having different sampling frequencies are to be coupled, the sampling frequency of one of the sources will have to be changed.
A method of sampling frequency conversion discussed, for example, in European Patent Application EP 0 052 847, consists of converting the digital signal to an analog signal and subsequently converting this analog signal to a digital signal having the desired sampling frequency. In order to avoid aliasing and all sorts of disturbing signals, the analog signal is to be carefully filtered. The filter, the digital-to-analog converter and the analog-to-digital converter make this method of sampling frequency conversion intricate and expensive.
Entirely digital solutions without the analog intermediate stage have been looked for. They are based on calculating, with the aid of digital interpolation and decimation filters, the value of new samples on the basis of available samples of the digital signal to be converted. As long as the given and desired sampling frequencies have a rational proportion, the conversion with the aid of such filters is effected in a relatively simple manner. The time patterns of the given and desired samples match in that case. Examples of such converters may be found in U.S. patent specifications U.S. Pat. Nos. 3,997,773, 3,988,607, 4,020,332 and 4,472,785. An instructive survey of the operation of interpolation and decimation filters may be found in the journal article entitled: "Interpolation and Decimation of Decimal Filters--A Tutorial Review", Proceedings of the IEEE, Vol. 69, No. 3, March 1981, pp. 300-331, which likewise provides extensive references to further literature.
It becomes more difficult if the time patterns do not match. In that case the distance between a desired and a given sample is always different. From European Patent Specifications EP 0 052 847, EP 0 099 600, EP 0 137 323, EP 0 151 829 and EP 0 227 172, sampling frequency converters are known in which the coefficients of an interpolation or decimation filter are constantly adjusted on the basis of this distance. With this type of converters extremely good results may be obtained, but they do require rather much hardware for calculating the coefficients.
A very simple manner of sampling frequency conversion is based on the omission or repetition of a specific number of samples from the incoming sample stream per time unit, depending on the frequency difference between the given and desired sampling frequencies. The process of sample omission, sample passing or sample repetition is termed validation. The validated sample sequence is rearranged with the desired sampling frequency so that an equidistant sequence of samples is obtained. This process leads to signal distortion. Omission/addition of samples from/to the original sample sequence and rearranging them according to a new time pattern results in expansion/compression of the digital signal. When the samples are rearranged according to the new time pattern, however, the extension/compression is eliminated again. However, an omission/addition results in a signal jump that had originally been absent in the analog signal. For a given sampling frequency difference, the expansion/compression has a fixed value causing a relatively ever enhancing phase shift for an increasing number of frequency components in the analog signal. In order to keep the analog signal distortion below a specific level, the sampling frequency is to be increased to a very high level, after which validation may take place, as is discussed in the article published in ICASSP 81. There it is stated that for an audio signal having a frequency of 20 Khz, which is to be sampled with a frequency of about 50 Khz, an oversampling factor 2.sup.15 =32768 is necessary for obtaining a signal-to-noise ratio of 90 dB after rearrangement. This necessitates an oversampling frequency of about 1.5 GHz. Such a high frequency renders the implementation of a sampling frequency converter difficult, intricate, expensive and hardly attainable with standard integrated circuit manufacturing processes.